SylkServer 2.3.0 release
SylkServer version 2.3.0 has been released with state-of-the-art cross-protocol SIP/XMPP multi-party conferencing capabilities. This is done by bridging MUC functionality from XMPP with MSRP-Switch functionality using SIP.
- SIP users can create ad-hoc conference rooms using SylkServer MSRP chat and invite users from both SIP and XMPP domains
- XMPP users can create multiparty conferences using SylkServer MUC capability and invite users from both XMPP and SIP worlds
Normative References¶
- XMPP Multi-User Chat http://xmpp.org/extensions/xep-0045.html
- SIP MSRP-Switch http://tools.ietf.org/id/draft-ietf-simple-chat-17.html
Changelog¶
- Added SIP/XMPP gateway ability to invite participants to a multiparty chat
- Added support for XEP-0030 (service discovery)
- Added RTP audio and MSRP chat 'echo' application
- Added timestamp to generated PIDF documents
- Improved logging when adding participants to a conference
- Improved logging for the XMPP gateway application
- Removed extended-away state handling as it no longer exists in the SDK
- Made several improvements to XMPP stanza parsing
- Fixed detecting MSRP Nickname collision
- Fixed handling presence stanzas without a resource part in the from
- Fixed translating resource IDs for presence
- Fixed leaking session objects if session fails while joining a conference
- Fixed mapping room URI for received REFER requests
The SIP/XMPP multi-party conferencing is documented here
Normative References¶
See SIP/XMPP Gateway documentation
Installation Instructions¶
Compatible Clients¶
The following clients have been successfully tested against this functionality: iChat, Jitsi, PSI, Adium (XMPP) and Blink Cocoa (SIP). Any client following same standards should be able to interoperate.
Zero Configuration¶
In the good tradition of 'just-works' there is nothing to configure, the functionality is made possible by just using drag in drop in the respective clients.
Live Service¶
SIP2SIP.info domain can be used as a multi-party inter-domain SIP/XMPP bridge. Just create a requested from either SIP or XMPP to username@conference.sip2sip.info.
Roadmap¶
Work is in progress for adding Jingle audio and File Transfer support.
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